Hi experts,
The company I work for is mexican. One of our customers has a branch in Miami and they have an AT&T SIP Trunk, the issue is that they can`t complete calls to Toll Free Numbers, I have called AT&T service and they said the problem is ours.
If I run a SIP debug in the gateway I receive the 403 SIP code forbidden just for Toll Free Numbers.
The gateway configuration is below.
service timestamps debug datetime msec
service timestamps log datetime localtime show-timezone
no service password-encryption
!
boot-start-marker
boot-end-marker
!
!
logging buffered 51200 informational
!
no aaa new-model
clock timezone MIAMI -5 0
clock summer-time MIAMI recurring 2 Sun Mar 0:00 1 Sun Nov 0:00
!
no ipv6 cef
ip source-route
ip cef
!
!
no ip domain lookup
multilink bundle-name authenticated
!
!
crypto pki token default removal timeout 0
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
address-hiding
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
session refresh
header-passing
error-passthru
registrar server expires max 3600 min 120
asserted-id pai
privacy pstn
no update-callerid
early-offer forced
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
voice class codec 2
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
voice class codec 10
codec preference 1 g729r8 bytes 30
codec preference 2 g711ulaw
!
voice class h323 1
h225 timeout tcp establish 5
!
voice class sip-profiles 1
request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
request ANY sdp-header Connection-Info remove
response ANY sdp-header Connection-Info remove
!
voice class sip-profiles 2
response ANY sip-header Allow-Header modify "UPDATE," ""
!
!
license accept end user agreement
license boot module c2900 technology-package uck9
hw-module ism 0
!
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!
redundancy
!
!
interface Embedded-Service-Engine0/0
no ip address
!
interface GigabitEthernet0/0
description To SIP PSTN
ip address 12.161.41.162 255.255.255.224
duplex auto
speed auto
!
!
interface Vlan1
no ip address
!
ip route 12.0.0.0 255.0.0.0 GigabitEthernet0/0
ip route 12.194.0.0 255.255.0.0 12.161.41.161
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
!
control-plane
!
!
mgcp profile default
!
sccp local GigabitEthernet0/1
sccp ccm 130.61.1.3 identifier 1 priority 1 version 7.0
sccp ccm 130.61.1.4 identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 222 register ConfMIA
associate profile 111 register XcodeMIA
associate profile 333 register MTPMIAMI
!
dspfarm profile 111 transcode
description DSP Resources Transcoding
codec g729br8
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 9
associate application SCCP
!
dspfarm profile 222 conference
description DSP Resources Conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 11
associate application SCCP
!
dspfarm profile 333 mtp
codec g711ulaw
maximum sessions hardware 1
maximum sessions software 4000
associate application SCCP
!
dial-peer voice 1000 voip
description llamadas locales Miami
preference 1
destination-pattern 3[0-9]........
session protocol sipv2
session target ipv4:12.194.99.53
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte cisco-rtp
ip qos dscp 5 media
no vad
!
dial-peer voice 5000 voip
description Toll Free Numbers
destination-pattern 18……...
session protocol sipv2
session target ipv4:12.194.99.53
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte cisco-rtp
ip qos dscp 5 media
no vad
!
sip-ua
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers trying 1000
timers connect 100
timers register 250
sip-server ipv4:12.194.99.53:5060
host-registrar
g729-annexb override
!
gatekeeper
shutdown
!
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 130.90.3.10 port 2000
max-ephones 58
max-dn 50
system message primary Altex USA Survivable Telephony
system message secondary AltexUSA Survivable
transfer-pattern 2..
transfer-pattern 3..
transfer-pattern 4..
keepalive 60
voicemail 6670
call-forward pattern #
call-forward pattern .T
call-forward busy 6671
call-forward noan 6670 timeout 16
moh GrupoAltexespanolOK.au
multicast moh 239.1.1.1 port 16384
time-zone 9
date-format dd-mm-yy
!
line con 0
exec-timeout 0 0
logging synchronous
login local
stopbits 1
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
logging synchronous
login local
transport input all
!
scheduler allocate 20000 1000
ntp server 130.61.1.10 prefer
end
Pleas Help me!!
Best Regards!
